Dynamic codec with priority for voice over IP in WLAN

Authors:
Kewin O. Stoeckigt;Hai L. Vu;Philip Branch
Affiliations:
Swinburne University of Technology, Melbourne, Australia;Swinburne University of Technology, Melbourne, Australia;Swinburne University of Technology, Melbourne, Australia
Venue:
MMSys '11 Proceedings of the second annual ACM conference on Multimedia systems
Year:
2011

Citing 4
Cited 0

Theory, Volume 1, Queueing Systems

Theory, Volume 1, Queueing Systems
Experimental tuning of AIFSN and CWmin parameters to prioritize voice over data transmission in 802.11e WLAN networks

IWCMC '07 Proceedings of the 2007 international conference on Wireless communications and mobile computing
VoIP codec adaptation algorithm in multirate 802.11 WLANs: distributed vs. centralized performance comparison

EUNICE'07 Proceedings of the 13th open European summer school and IFIP TC6.6 conference on Dependable and adaptable networks and services
SP 800-58. Security Considerations for Voice Over IP Systems

SP 800-58. Security Considerations for Voice Over IP Systems

Quantified Score

Hi-index	0.00

Visualization

Abstract

Research on Voice over IP (VoIP) in infrastructure wireless local area networks (WLANs) has shown that, because of the access mechanism, the access point severely limits the number of voice calls a WLAN can support. To address this problem we propose a new scheme based on the IEEE 802.11e quality of service (QoS) mechanism where a tradeoff between codec quality and priority is exploited to improve the number of calls that can be supported. In particular, we propose certain priority settings at the access point (AP) to encourage users to switch to a lower quality codec during periods of high contention and thus enable them to maintain the call. We develop a detailed analytical model to show the benefit of this scheme. Our analytical results are validated by extensive simulation and show that the voice capacity in the network can be significantly improved. Furthermore, by using the ITU-T E-model to assess the voice quality, we show that users with a lower quality codec can still maintain an acceptable level of quality using the proposed scheme.